I might not be able to configure sipx the way I want, but been working in
sales and system-consulting for 4 years now.
-breakdown of business telephony is major business killer
-according to Murphy your T1 WILL break down as soon as you rely on it too
much
-rerouting to mobile is great, but what
Nathaniel,
It really needs to be an analysis of how that branch functions. Depending
on the role the branch serves within the organization in the event of any
natural disaster, especially for intra-office communication, relying on a
T1 to call down the hall, upstairs, downstairs, etc, is perhaps
What Tony says is definitely worth thinking about. What would you do if the
T1 was down for a week or two?
A gateway out at the remote for 911 / backup at a minimum.
The Patton Smartnode gateways are functional enough that you could have a
secondary line on certain/all phones registered
Hi!
I think i have read all there is...but upgrading using yum fails.
It is a working 4.0.4 system and i have checked all repo details
and pasted urls into browser. All files are there.
A yum update complains about sipx-freeswitch not beeing there.
A rpm -qa shows it is.
Clues?
Thank you.
..
Ah, but ona regular sipx installation the only freeswitch package installed
by default is
sipx-freeswitch
[r...@sipx ~]# rpm -qa sipx-freeswitch
sipx-freeswitch-1.0.3-1
spidermoney is installed, but tis not used in 4.0.x or 4.2.x as I understand
it.
So you have a non-standard install and need
Thank you, i dont think so nonstandard but
initially a 4.0.2 from iso, upgraded to 4.0.4.
I removed all packages it complained about with rpm -e no complaints
there.
That worked. :-)
*Vänliga Hälsningar/Best Regards*
2010-07-01 11:49, Tony Graziano skrev:
Ah, but ona regular sipx
Just wondering how they got there to begin with...
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract
Hi all,
looking at
http://wiki.sipfoundry.org/display/xecsuserV4r2/Custom+FreeSWITCH+programming i
see that it's quite possible to use local freeswitch as sip - h323 gateway.
I'm going to implement it on the test sipx 4.2.1 instance first. I use sipx
installed from 4.2.1 (ezuce) iso.
I found
The remote contains these offices (I'm not sure what some of these offices due
- some are State/Federal offices):
Public Utilities
Permits Inspections
Extension Office
NRCS GSCD
Farm Service Agency
Rural Development
Election Board
I totally agree that putting all your eggs in one T1
I tried to find the perfect combination of centralized/decentralized for a
similar setup. I wanted centralized for the same reason you wanted. I'm in
healthcare, and we could not have any potential issues with the remote site
dialing out if the T1 to our main office went down. In the end, I
On Thu, Jul 1, 2010 at 8:08 AM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
The remote contains these offices (I’m not sure what some of these
offices due – some are State/Federal offices):
Public Utilities
Permits Inspections
Extension Office
NRCS GSCD
Farm Service Agency
Well,
try to think all possibilities through.
I don´t think its a matter of telephony platform at all, but a matter of
availability , ROI and
So on.
I don´t know about US-broadbandaccess-technology but in the scenario
described by you in germany
I would strongly advice for redundant
It's probably simpler in his case...
He needs to look at each of the different ways to connect a remote site (and
he knows what they are already), and decide if T1 goes down what happens
with the users on that link? How will they know it is down, what is the
procedure to find out when it will
I am installing a sipx system with audiocodes FXO gateways MP114and MP118
outbound dialling is working, but incoming calls are answered then returns
dial tone , I can then dial 0 and get to the operator and then dial my
extension numbers
Thanks
Bob Anderson
Cyrand Corp
3-4170
Might I suggest putting a redundant proxy out at the remote location and
configuring your DNS like this:
http://wiki.sipfoundry.org/display/xecsuser/Location+based+DNS+views+for+sipXecs+using+BIND
so all phones at that location register to that redundant proxy.
From experience I would get a
Might I suggest a private MPLS cloud instead of the T1's with tagging for
voice... this way he can use VVX's out on the fringe and we all know how
fond of those he is.
On Thu, Jul 1, 2010 at 9:15 AM, Josh Patten jpat...@co.brazos.tx.us wrote:
Might I suggest putting a redundant proxy out at
I've never done anything like that before, but I guess it sounds like a
winner...
In any case 1 T1 isn't going to cut the mustard when running 40 phones
and that many departments. I have a site with 6 people and 7 phones and
I had to put a secondary proxy out there because the signalling
Still a bit new to the technology, but I believe I had exactly the same
issue.
It seems that the autoconfiguration part in 4.2 is not quite right for the
AudioCodes. Log on to the AudioCodes device, and look in the autodial
section of the AudioCodes device's configuration... it's probably blank
The audiocodes firmware is 5.8
Cyrand Corp
3-4170 Sladeview Crescent
Mississauga, ON L5L 0A1
905-817-8208 x237
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List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe:
It stands for Multiprotocol Label Switching. It’s a way of speeding up
network traffic by avoiding the time it takes for a router to lookup the
address of the next node to send a packet to. In an MPLS network data has a
label attached to it, and the path that is taken is based on the label.
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of m...@grounded.net
[m...@grounded.net]
I didn't even think and didn't even remember that. My first thought was to ask
if this was something others had seen
You will also need to look at how your ITSP handles 911/999 emergency
calls. If the 'remote' location has a separate emergency center to
receive 911/999 etc. you will need to address that.
Regards,
Don McIlvin
Telecom Department
Realogy/NRT/Coldwell Banker
52 Second Ave. (3rd Floor)
Waltham, MA
Its easy enough to do with a local analog gateway. Without registering the
phone can send calls to the gateway. That should be a no brainer by now.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
I think that when planning setup with remote locations one may want to take
a look at AuidioCodes SAS (Stand Alone Survivability) functionality.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Tony Graziano
I think we covered the fact that the phones can register to the gateway as a
failsafe also. Patton or AC.
On Thu, Jul 1, 2010 at 12:17 PM, Nikolay Kondratyev k...@nstel.ru wrote:
I think that when planning setup with remote locations one may want to take
a look at AuidioCodes SAS (Stand Alone
I think the T-1 will give him a better network choice, as you wont have to
deal with the added latency of MPLS. A simple router can provide adequate
QOS for Voice and Data connections, and the cost of a dedicated T-1 from the
Teleco in most areas is very affordable these days, especially for
A private mpls network would introduce very minimal latency. But I think I'm
done on this thread. the fact that he has that many users on a T1 is going
to introduce latency when it is heavily used. MPLS may not be an option for
him depending on how many sites he has like this. I recall Nathaniel
Oh, there are such things as private MPLS networks, which is what i
suggested. That would beat the pants off a T1 or DS3, directly connecting
layers 23 while giving a layer 2 experience. It might be different if it
were an internet port also, but while that is a choice, MPLS labels
things like
Hi!
Finally got this working with sipx.
Ditched sendmail
Installed postfix/sasl2/certs etc.
Took quite a while.
Now relaying mail via outside provider works.
From what i could tell on the wiki people
has found this less than fun to configure.
If anyone is interested i can mail links, main.cf
OK I've tested the 18944 sipxregistry RPM douglas built yesterday and
neither hunt group pickups NOR alias pickups are working. Please see
http://track.sipfoundry.org/browse/XX-8438 for more details and a snapshot.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979)
BTW for those that didn't think this functionality ever existed,
http://track.sipfoundry.org/browse/XX-4775 read the description.
Also, I've confirmed hunt group pickup and alias pickup don't work in my
EDE environment either (currently running build 18948)
Josh Patten
Assistant Network
Is there an easy way to configure a group invite to a conference room ?
Warm Regards
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Centre Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-693-2833
Fax: 615-628-5698
Sorry - thought I'd sent this already...:
Tony - as usual, speaketh the truth.
Here is what I'm leaning towards (at least for the next few minutes - or until
someone makes a compelling case to do otherwise):
1) Have multiple weighted routes to the primary location (T1 as the
default and
You haven't addressed the overtaxed T1 circuit though. I guarantee
you'll have problems if you try to ride all that traffic over 1 T1.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/1/2010 3:49 PM, Nathaniel Watkins wrote:
Sorry -- thought I'd sent
We've been doing some traffic logging on the T1 - we easily have 1 meg
available at any given time (and will of course be doing qos). I'm going on
the basic assumption that we will need to support 10 concurrent voice
connections over the T1. This 'remote' is about 5 miles away from our main
Sorry - I'm a bit behind on my mailing list today. I just sent an email that
shows that the T1 isn't overtaxed at the moment. We really only have 15-20
people that would use this for internet/email - many of them are inspectors
and/or are in the field and don't get email or browse the web.
Ok if you're sure it will work go ahead. But be prepared to fork over
for another T1 or a different solution altogether (such as MPLS) just in
case things don't work out.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/1/2010 3:52 PM, Nathaniel Watkins
Sure it will work...ha...famous last words...
Initially this was for 20 phones, which I was pretty confident about -
45...ummm, we'll see what happens. I'll have a much better feel once I put the
first 20 on...they are the ones with most of the call volume...so we'll see.
I'll take it slow
On 7/1/10 4:37 PM, Gabe Casey wrote:
Is there an easy way to configure a group invite to a conference room ?
might be something different in different terms.
What do you want to invite to what? what kind of group? a group of
people? phones? im contacts? hunt group? park group? what?
are you
After thinking about it, I would like to know if there is a way of removing a
single reply from the list. Not sure why/how it went to the list since I know I
sent it personally, at least it isn't the list as the email address. I don't
mind the list seeing the reply if that is the case but I
Yes Michael, a sip invite to a group of extensions. Similar to your ability to
invite a single user from the Web UI which invites a participant to the
conference from the conference extension.
Perhaps someone has solved this via the SOAP api or using paging groups. Just
looking for best
My guess is you are out of luck - there are a bunch of sites that grab a copy
of everything on these lists for future generations...
On a side note - which one are you talking about - I'm sure you've now peaked
everyone's interest...sorry, being counterproductive now.
-Original
You should be able to use a curl request via php. Here is the meat of the code
– you should be able to loop thru this for each person you want to invite:
?php
$to=$_GET['to'];
$from=$_GET['from'];
$pass=$_GET['pass'];
$url =
On Thu, 1 Jul 2010 17:57:52 -0400, Nathaniel Watkins wrote:
My guess is you are out of luck - there are a bunch of sites that grab a
copy of everything on these lists for future generations...
Not cool, I know I didn't send it public, I still have the outgoing headers of
the original showing
Thanks Dale - I had to forward this to the group :)
Now I'm going to have to lookup 'pedant'...
I didn't know there were any coders (making an assumption here...) that knew
how to spell. I also learned something today...
~Pore Spellier
-Original Message-
From: WORLEY, Dale R (Dale)
Hmmm I would say that its not all that easy considering the first person who
answers will stop the call. Maybe an email or calendar invite with the threat
of job loss.
Even external users should be able to dial in. If its cost maybe use toll free
or ISN dialing
-Original Message-
Ok. Multiple outbound calls. Works even better.
-Original Message-
From: Nathaniel Watkins nwatk...@garrettcounty.org
Sent: Thursday, July 01, 2010 6:02 PM
To: Gabe Casey gca...@franklinamerican.com; Michael Scheidell
scheid...@secnap.net
Cc: sipx-users@list.sipfoundry.org
A pedant is a person who is overly concerned with formalism and precision.[1]
Make that 2 things I've learned today...
Might I add - 2 characteristics of a good coder...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On
On Thu, 1 Jul 2010 18:09:03 -0400, Nathaniel Watkins wrote:
A pedant is a person who is overly concerned with formalism and
precision.[1]
Gee, and that's how all of this got started with to begin with hehe.
___
sipx-users mailing list
Ahh - it's great when a post comes together so nicely.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of m...@grounded.net
Sent: Thursday, July 01, 2010 7:13 PM
To: sipx-users
Subject: Re: [sipx-users] Removing a
Did you get yours in yet? I'm not finding anything for SRV with these.
On Sun, 27 Jun 2010 21:07:18 -0500, Josh Patten wrote:
http://www.grandstream.com/products/gxv_series_phone/gxv3140/gxv3140.html
I've
got one of those on order to test with. I'll let everyone know how it
goes. once I get
FYI: http://track.sipfoundry.org/browse/XX-8474 check the diagram and
the bottom comment.
My coworker (the developer) has a beta of this fix written and we are
testing it now in our EDE environment. Everything is working so far but
we want to test every aspect of it to make sure it is stable
This never touches the SIP stack, it only deals with parsing of the
resource list XML and changing values in the internal database of
subscriptions.
On 07/01/2010 05:45 PM, Todd Hodgen wrote:
Was this new approached checked with compliance of the necessary standards.
That's my only thought.
Excellent Nate ! Thanks for the suggestion. I think that would do the trick.
Warm Regards
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Centre Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-693-2833
Fax: 615-628-5698
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